Communications system and related method for automatic call distribution

ABSTRACT

In one non-limiting example, the system and method for communicating establishes a Session Initiation Protocol (SIP) communication session call between a caller and a communications server. The call from the caller is queued at the communications server. A first SIP Invite command is sent between a call agent station and the communications server to dequeue the call. A second SIP Invite command is sent between the call agent station and the communications server. The call is transferred to the call agent station to establish a communications session between the caller and the call agent station.

FIELD OF THE INVENTION

The present invention relates to communications systems, and moreparticularly, this invention relates to communications that use theSession Initiation Protocol (SIP).

BACKGROUND OF THE INVENTION

A communications server such as an Automatic Call Distributor (ACD)often queues a caller while the caller waits for live support. Somesystems include a Private Branch Exchange (PBX) having circuitry andmodules that allows the system or agent to manually dequeue one of thesecalls. Similar features occur in Private Branch Exchange (PBX) systemsthat generally offer a feature for call pick-up. This call pick-upfeature allows one user to retrieve a call destined to a remote user asdirected to or within a predefined group of users as a group. In both ofthese scenarios when Session Initiation Protocol (SIP) phones areinvolved, however, the media stream must be negotiated in an end-to-endfashion, without violation of the offer/answer model described inRFC3264, the disclosure which is hereby incorporated by reference in itsentirety. This should not involve extra effort on the user's part, suchas having to answer the phone after making the request.

SUMMARY OF THE INVENTION

In one non-limiting aspect, the dequeing endpoint is able to make a callto an extension that is then terminated while the queued call istransferred to the dequeing endpoint. This transfer of the callprocesses all necessary SIP Invites with or without the SessionDescription Protocol (SDP) to negotiate the appropriate mediaparameters. The system then forms a call to the dequeing endpoint usinga ring type that allows the call agent station as a typical SIP phone togo off hook to retrieve the call either to a handset or speaker, withoutinteraction of the user in what is termed a Hands-Free Auto Answer(HFAA).

In one non-limiting example, the system and method for communicatingestablishes a Session Initiation Protocol (SIP) communication sessioncall between a caller and a communications server. The call from thecaller is queued at the communications server. A first SIP Invitecommand is sent between a call agent station and the communicationsserver to dequeue the call. A second SIP Invite command is sent betweenthe call agent station and the communications server. The call istransferred to the call agent station to establish a communicationssession between the caller and the call agent station.

In one example, the call from the communications server to the callagent station is transferred without user interaction. The call is alsodequeued by dialing an extension from an SIP phone at the call agentstation and sending the SIP Invite command as a dequeing operationcorresponding to a Session Description Protocol (SDP) offer andnegotiating new and current media parameters. These new and currentmedia parameters are negotiated using an SIP Invite command offerwithout any SDP media parameters to gain the new and current mediaparameters. In another example, it can be accomplished with the SDPmedia parameters.

In yet another example, the media is transferred to the call while thecall is queued using the real time transport protocol (RTP). The SIPRefer command is used in another example for transferring the queuedcall to the call agent station. In still yet another example, a dialableextension is called from the call agent station to dequeue the call.This call agent station in an example is an SIP phone and can include aspeaker phone. The communications server is typically formed as anAutomatic Call Distributor (ACD).

BRIEF DESCRIPTION OF THE DRAWINGS

Other objects, features and advantages of the present invention willbecome apparent from the detailed description of the invention whichfollows, when considered in light of the accompanying drawings in which:

FIG. 1 is a high-level flowchart showing a method that can be used inaccordance with a non-limiting example such as applied with acommunications server that queues a call waiting for connection to acall agent station.

FIG. 2 is a high-level block diagram showing the communications systemhaving a communications server and other devices through which themethod shown in FIG. 1 can be applied in accordance with a non-limitingexample.

FIG. 3 is another high-level flowchart showing an example method asapplied with a communications server as a Private Branch Exchange (PBX)in accordance with a non-limiting example.

FIG. 4 is a high-level block diagram showing a communications systemhaving a communications server and other network devices through whichthe method shown in FIG. 3 can be applied in accordance with anon-limiting example.

FIG. 5 is an example of a communications system that can use the methodas described and incorporate components of the communications systemsshown in FIGS. 2 and 4 in accordance with a non-limiting example.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

The present invention will now be described more fully hereinafter withreference to the accompanying drawings, in which preferred embodimentsof the invention are shown. This invention may, however, be embodied inmany different forms and should not be construed as limited to theembodiments set forth herein. Rather, these embodiments are provided sothat this disclosure will be thorough and complete, and will fullyconvey the scope of the invention to those skilled in the art. Likenumbers refer to like elements throughout.

A communications server such as an Automatic Call Distributor (ACD) willqueue callers that are waiting for live support. Some systems provide afeature that allows a call agent to dequeue manually one of these calls.When Session Initiation Protocol (SIP) phones are involved, the mediastream should be negotiated in an end-to-end fashion without violationof the offer/answer model described in the incorporated by referenceRequest for Comments (RFC) 3264. This should be accomplished withoutinvolving extra effort on the user's part, such as having to answer thephone after making the request.

When initiating multi-media teleconferences, voice-over-IP (VoIP) calls,streaming video, or other media sessions, the media details, transportaddresses and session description metadata are conveyed to participantsusing the Session Description Protocol (SDP) as the standardrepresentation for this information. SDP is used irrespective of howthat information is transported. SDP is a format for the sessiondescription, but does not incorporate any transport protocol. SDPconveys information about media streams in multi-media sessions to allowthe recipients to participate in the session. SDP describes streamingmedia initialization parameters in an ASCII string in terms of a sessionannouncement, session invitation, and parameter negotiation. It is ashort, structured text description and conveys the name and purpose ofthe session and the media, protocols, CODEC formats, timing andtransport information required for the session.

The Session Initiation Protocol (SIP) is an application-layer controlprotocol that creates, modifies and terminates sessions. The SIPmessages create sessions and carry session descriptions that allowparticipants to agree on a set of compatible media types. These sessiondescriptions are typically formatted using SDP. When used with SIP, theoffer/answer model as set forth in RFC 3264 provides the framework fornegotiation using SDP. The SDP includes a session name and purpose, thetimes the session is active, the media comprising the session, andinformation used to receive those media, for example, the addresses,ports, and formats. Entities use SDP to arrive at a common view of themultimedia session between them. In the offer/answer model, one entityoffers the other entity a description of the desired session from theirperspective, and the other entity answers with a desired session fromtheir perspective. SIP uses this offer/answer model as its negotiationframework.

In the offer/answer model, a party indicates a desired sessiondescription from their point of view as the SDP offer, which containsthe set of media streams the offerer wants to use, the desiredcharacteristics of the media streams as qualified by a format parameterand media-line attributes, the IP addresses and ports where the offererwants to receive the media, and additional parameters, if required, thatqualify the media transport. The other party receives the offer andreplies with an SDP answer accepting or rejecting the media stream. Forexample, if a media stream is not accepted, the port value in an m-linefor media is set to zero. An answer typically includes the mediastream's characteristics used for the session and the IP addresses andports that the answerer wants to use to receive the media. It should beunderstood that the offer and answer is an “atomic” entity and theremust be one of each. It must be negotiated every time. It can be thesame information as a previous offer/answer. There still must be activenegotiation.

In accordance with a non-limiting example, a dequeing endpoint is ableto make a call to an extension that is terminated while the queued callis transferred to the dequeing endpoint. This transfer of the callprocesses all necessary SIP Invites with or without the SessionDescription Protocol (SDP) media parameters to negotiate the appropriatemedia parameters. It then forms a call to the dequeing endpoint using aring type that allows the SIP phones to go off-hook (called Hands-FreeAuto Answer) to retrieve the call either to a handset or a speaker,without interaction of the user that made the call.

FIG. 1 is a high-level flowchart showing an example of a method that canbe applied in accordance with a non-limiting example. The callerinitially makes the call that requires usually an agent to assist thecaller such as ordering a product or other function (block 10). Adetermination is made if the agent is available (block 12) and if yes,then the caller is connected to the agent (block 14). If not, thecommunications server, such as the Automatic Call Distributor queues thecall (block 16) such as in a queueing module or other memory device ofthe communications server. This communications server could also be aPrivate Branch Exchange (PBX) in a non-limiting example.

At this time a call agent station, such as having a live agent, dequeuesthe call typically by dialing an extension and making the first SIPInvite (block 18). Negotiation occurs for typically new and currentmedia parameters and a second SIP Invite is made as a Hands-Free AutoAnswer (HFAA) (block 20). The call is transferred to the call agentstation (block 22), and the live agent receives the call and thecommunications session is established between the caller and the callagent station, i.e., typically the live agent. The process ends (block24).

FIG. 2 is a high-level block diagram of a communications system 40showing the various sequences between the caller 42 that makes the call,such as to obtain a live agent, and the call agent station 44. A call isinitially established as an SIP call between the caller and thecommunications server 46 as an ACD in this example using the Invitecommand and an OK response. Music is transferred while the caller isheld in queue 48 using the Real-Time Transport Protocol (RTP), whichdefines a standardized packet format for delivering audio and video overthe internet. The “live” call agent at the call agent station 44 desiresto dequeue the caller from the communications server as the ACD 46 andissues an Invite command. This is followed by OK back and Bye asillustrated. The Invite command is a call to an extension. The callagent at the call agent station 44 will typically dial four numbers asthe extension for this dequeing operation. An Invite command as aHands-Free Auto Answer (HFAA) is established as an alert and an OKestablished. The SIP Refer command is initiated to transfer the callbetween the caller and call agent, as illustrated.

It should be understood that the media in this particular example isestablished in the communications server as the ACD. In some techniques,a call agent could have a piece of software running on a personalcomputer, and the call agent in that example pushes a button to route.If the call agent at the call agent station does not have thatfunctionality, however, and only has an SIP phone as in this example,then the call agent is limited in function. The system and method, inaccordance with a non-limiting example, solves this technical problem.The system and method in one example creates a dialable number with theagent at the call agent station 44. When the call agent calls thedialable number, it is invoking the dequeing operation. The call isdisconnected and the transfer occurs as noted above. The system andmethod further accomplishes a dialable number as the double SIP Invite(also termed Reinvite) as illustrated between the call agent andcommunications server as the ACD and the invoking of the SIP transfer.Internally, a Refer (XFER) command as an SIP transfer is initiated. Thissystem and method as set forth is advantageous such that the call agentdoes not call the ACD and connect to a “box” as a media server in oneexample where both the agent and caller negotiate a call. This takesadditional steps and processing.

The system and method as described has less messaging and ultimatelyremoves the ACD 46 from the call signaling. If there were a number ofmedia reinvites, this could create calls from both sides and create moreoverhead to the communications server as the ACD. The system and methodas described is more efficient in the processing power and networkusage.

FIG. 2 as described before is a high-level block diagram showing basiccomponents that can be used with the method as described relative to theflowchart in FIG. 1. The caller can use typically any type of calldevice such as a telephone that makes the call to the number of eitherseries of prompts or call agent, for example, when ordering a product.The call agent station 44 typically includes at least one live agent(and could include a group of live agents). The call is routed andtransferred to the communications server 46 through a communicationsnetwork 50 as an illustrated IP network in this example, but could beany network. The communications server 46 in this example is anAutomatic Call Distributor, and includes functionality such as supportedby a NetVanta 7000 series or 7100 series device, including PrivateBranch Exchange functionality such as manufactured by ADTRAN, INC. ofHuntsville, Ala.

The communications server 46 includes communications ports 52 and aprocessor 54 and transceiver 56 that communicates and processes datapackets for data communications and a database 58 for storing data. Themedia server 60 is included in this example and processes media andother data and interoperates with the other components. The call agentstation 44 can include a plurality of different network devices such asan SIP phone or other network device and, in this example, the devicetypically includes a processor 70, transceiver 72, database 74 andcommunications ports 76. A number of live agents could be situated atthe call agent station 44. These live agents could have differentnetwork devices and interspersed and geographically spaced networkdevices. As described before, the different data flows indicative of theSIP data transfer and negotiations are shown, including various SIPcodes in this non-limiting example. For example, the Invite between thecaller and communications server is an Invite (2200) and the Okay is acode 200. The Invite between the communications server 46 and the callagent station 44 is an Invite (2205) as a dequeue operation.

The communications system and method is described relative to FIGS. 1and 2 as a communications server and as an Automatic Call Distributor,but the basic methodology is applicable with a Private Branch Exchange(PBX) type of system. These types of systems generally offer a featurefor call pick-up that allows one user to retrieve a call destined to aremote user as a directed call or within a defined group of users as agroup. When SIP phones are involved, the media stream in one example isnegotiated in an end-to-end fashion without violation of the offeranswer model described in the RFC3264, and without involving extraeffort on the user's part, such as having to answer the phone aftermaking the request.

In accordance with a non-limiting example, the system and method createsa forwarded call from the initial calling party to the pick-up requestorand uses a ring type that allows the SIP phone to automatically gooff-hook as a Hands-Free Auto Answer (HFAA) to retrieve the call eitherto the handset or the speaker.

FIG. 3 is a high-level flowchart illustrating an example method using aparty A at a first call device that calls party B at second call deviceand party C at a third call device. Party C monitors party B in thisexample, such as being in the same office, next door, or part of adefined group.

In this non-limiting example, party A at the first call device callsparty B at a second call device, while party C at a third call device ismonitoring party B (block 100). A determination is made if party B picksup or answers (block 102), and if yes, the call is established betweenparty A and party B for a communications session (block 104). If not,the system determines if party C is monitoring party B such as throughan indicator or other means (block 106), and if not, the call isterminated or party A may leave a message such as through prompting(block 108). If yes, then party C dials an extension for a first SIPInvite command (block 110). This first SIP Invite command is sent fromthe third call device to the communications server as a pick-up call andinstruction for the call to the second call device when it has notanswered the call from the first call device. A second SIP Invitecommand is sent between the third call device and communications server(block 112). The call is transferred from party B as the second calldevice to party C as the third call device to establish a communicationssession between party A and party C as first and third call parties(block 114). The process then ends (block 116).

As indicated before, party C is monitoring party B and notices the ringafter party B receives the call and the SIP device rings. Thismonitoring could be by hearing the ring as an audible indicator (abuzzer or phone ring) or a digital indicator such as a blinking light orother means located at party C. The call from party A was originallyinto the server as a Private Branch Exchange (PBX) in thiscommunications server example and the call routing could be through atelecommunications network as a cellular network, PSTN or any other typeof communications network into the PBX. Party C could be located at adesk next to party B in this particular example, such as close workingcoworkers. Party B may not currently be present or could be located inparty C's office. In this example, party B is not able to answer his orher SIP phone and party C is aware of that fact. Party C notices the SIPphone of party B ringing either through the audible indicator or digitalindicator in a non-limiting example. Party C now desires to pick up thecall and take it from party B. To accomplish that, party C invokes thesystem and method as described. It should be understood that party B andparty C could be part of a defined group and the call from thecommunications server as the PBX could be directed generally to thegroup, which includes parties B and C.

FIG. 5 shows a communications server 146 that is connected to a firstcall device 120, a second call device 122, and a third call device 124as explained before which includes respective party A, party B and partyC at the respective first, second and third call devices. Thecommunications server 146 in this example is a call control agent as aPBX in a non-limiting example and includes basic components as describedbefore (and given reference numerals in the 100 series) such as theports 152, processor 154, transceiver 156, and database 158 and couldinclude a media server 60 and Queue module 148. The call agent station124 as the third call device could include typical components such asdescribed relative to the call agent station of FIG. 2, such as theprocessor 170, transceiver 172, database 174 and ports 176. The call ismade between party A and party B as illustrated and the call istransferred to party B through the communications server 146. The callis cleared. Party C understands that party B is not available andperforms the pick-up call for party B using the first Invite (althoughnot illustrated in detail) and the response as a pick-up group callresponse with the session terminated. The Hands-Free Auto Answer (HFAA)is made that includes the double Invite as described before such thatthe call is then transferred and the communications session establishedbetween party A and party C.

SDP is a format typically used for describing the streaming mediainitialization parameter in an ASCII string. It describes the differentmultimedia communication sessions, including announcement and invitationand negotiation. SDP commands are used for negotiation between theendpoints of media types and not for communicating the data itself. Thedata is transported in a packet format in accordance with the Real-TimeTransport Protocol (RTP). RTP is a standardized packet format fordelivering audio and video over an internet and involves typicallystreaming media. The packets can carry media streams controlled by H.323and the Session Initiation Protocol (SIP) signaling protocols. RTPallows end-to-end, real-time transfer of the multimedia data.

In RTP, the sequence number is typically 16 bits and increments by onefor each RTP data packet while the timestamp reflects the samplinginstant of the first octet in the RTP data packet. By implementing thesession and delivering media to the user in this packet format, it ispossible to reuse the same network socket as the source IP and portnumbers and update RTP timestamps and sequence numbers and transfer themedia to the user with minimal jitter buffer flushes and networkresource usage.

The media server 60,160 as shown in FIGS. 2 and 4 at the communicationsserver is “on the box,” i.e., integrated with the communications server,instead of external to that device, but it could be external. Anytimestamps and sequence numbers can be negotiated for packets. Any audiois passing through RTP streams with the updated sequence numbers andtimestamps to make consecutive across any switchovers. It should beunderstood that buffering can be in the phone at the caller (phone).

In one non-limiting example, the system can integrate media servers tothe communications server of a number of different media servers thatare internal to the communications server. Thus, it is possible to useelements in C data structures passed as a message between internal mediaservers. When switching between media servers, the system passes theinformation about the last RTP packet sent to the caller by the previousmedia server. The system knows where the new media server should startany of its timestamps and sequence numbers. The system passes the lastRTP timestamp, sequence number and the real time of the last packetsent. Because it knows the actual time the last packet was sent, the newmedia server can determine how much to increment the last timestamp itwas provided. Any sequence number is easier to deal with because it issimply incremented and does not have to take the passage of timeinvolved in the switchover of media servers.

There now follows a general description of a larger IP phone network asa general description to show a more specific and larger network exampleto which the example can be applied. FIG. 5 is a system diagram of anInternet Protocol (IP) telephone system 200 that includes variousnetwork components and devices as shown in FIGS. 2 and 4 and otherinterconnected platforms, switches and servers. It should be understoodthat the system 200 shown in FIG. 5 is only one non-limiting example ofan IP telephone system that uses an example network device ascommunications server 210 for different functions, including a switch.This series of devices can include a NetVanta 7000 or 7100 device asmanufactured by ADTRAN, INC. of Huntsville, Ala. In one aspect, theNetVanta 7100 is a communications server as an all-in-oneoffice-in-a-box that provides voice and data solutions, includingPrivate Branch Exchange (PBX) functionality.

FIG. 5 shows the communications server 210 (such as the NetVanta series7000/7100) connected to an IP network 212, which uses SIP communicationsto various remote sites (A-C), which each include other devices thatcould operates as a communications server. Each of the remote sitesincludes a network device 214, 216, 218 of remote sites A, B and Crespectively, that operate as communications servers such as IP businessgateways. For example, remote site A could include an IP gateway device214 such as the Total Access 900 manufactured by ADTRAN, INC. ofHuntsville, Ala. Remote site B could include a NetVanta 6355 as an IPgateway device 216. Remote site C could include another communicationsdevice 218 similar to that at the host site, which includes a NetVanta7100 device. These are only examples for purposes of description. All ofthese communications devices operate as network switches and includeother functionality. For example, the device 214 at remote site Aconnects to a key system and various analog phones through acommunications system and an SIP phone. The device at remote site A alsoconnects to a PSTN. The device 216 at remote site B connects to the PSTNand an SIP phone, a PC/soft phone, a laptop and an IP printer asillustrated in that network. The device 218 at remote site C connects tothe PSTN and an SIP phone, a computer and a server. The host siteconnects to the PSTN and also SIP phones, computers and a server throughvarious communications connections and ports as illustrated.

There now follows a more detailed description of the communicationsserver 210 as shown at the host site and described as a NetVanta 7000series, and in this particular example, as a NetVanta 7100 for purposesof general understanding and description. This description can apply toother devices at the other remote sites.

In this device as the communications server 210, a single chassis canprovide a LAN-to-WAN infrastructure and Quality of Service (QoS) thatmaintains voice quality and includes a Graphical User Interface (GUI)for network set-up and facilitate installation and systemadministration.

In this example, this communications server can allow a converged IPvoice and data network with a full-function IP PBX for voice. It caninclude an integrated Power Over Ethernet (POE) switch-router for datain an integrated device and a Virtual Private Network (VPN) for secureinternet tunnelling. The device enables VoIP by providing theappropriate functionality that includes SIP-based telephony features,voice mail, multi-level auto-attendant, caller ID name/number, and otherfeatures for a complete VoIP network. The device includes multi-site SIPnetworking and SIP trunking service. Various optional modules couldinclude T1 and ADSL Network Interface Modules (NIMs). Analog (FXS, FXO)Voice Interface Modules (VIMs) can be included with T1, PRI voiceinterface modules and fiber SFP modules.

The communications server 210 in this example is an integratedcommunications platform and includes capability of a fast Ethernetswitch with Gigabit uplinks and expansion slots for the networkinterface modules and voice interface modules. The IP telephone system200 as illustrated includes voice mail and multi-level auto-attendant,caller ID name/number, COS, trunk groups, music-on-hold, sales-on-hold,overhead paging, and other call options, including call coverage lists,forwarding of calls to a cell phone and email notification of voicemail. The device can operate as an integral SIP gateway with theappropriate FXS and FXO analog interfaces to support analog phones, faxmachines, modems and credit card readers. The integrated voice mail caninclude 3,000 messages on eight ports and multi-level auto-attendantthat are multi-level on eight ports in the example of the NetVanta 7000series. The device includes, in one example, a full function IP accessrouter and an integrated state inspection firewall protects against theDenial-of-Service (DOS) attempts. The devices include IP Sec VP andtunnelling with DES/3DES/AES encryption and an SIP-aware firewall, andinclude T.38 support and a door relay, music-on-hold (MOH) interfacesand Voice Quality Monitoring (VQM).

SIP networking is supported between multiple locations such that abusiness can connect multiple sites and have three or four digit dialingand local call routing and survivability and on-net calls for tollbypass. Multiple SIP trunks allow one communications server to connectto other communication servers, such as a Total Access series device asmanufactured by ADTRAN, INC. Up to ten or more remote SIP trunks can besupported and connect to all endpoints at all locations such that a usercan have local voice mail and auto-attendant services. A hub and spokeSIP network can be accomplished in another example. A dedicatedcommunications server such as shown in FIGS. 2 and 4 and also used inthe IP phone system of FIGS. 2 and 4 can aggregate SIP trunks at acentral location, which for qualified applications, increases the numberof other communication servers that can be networked together.

The user can use an Internet Protocol (IP) phone such as an IP 700series of telephones with six or twelve line versions and supportmultiple call functions. It is possible to incorporate voicemail-to-email applications and personal auto-attendant in which eachphone sets up their own automatic attendant. It is also possible for thecommunications server to ring a series of stations and one externalphone number. The server can include a PC-based phone manager and it ispossible to incorporate an Internet Protocol (IP) soft phone to enableVoIP communications from a Windows- or Vista-based laptop or desktop PC.Through a PC-based phone manager, a user can customize phone settings.

It is also possible for the communications server to work in amulti-vendor environment and with an integrated T1-PRI trunk toconsolidate separate voice lines and internet access onto a single T1 orPRI trunk. It is possible to combine the IP and analog communicationsand support analog trunks, analog phones, fax machines and credit cardreaders without the requirement for analog telephone adaptors. It isalso possible to provide always-on, voice, data and high-speed dataaccess to business resources from a remote home office using a singlecable or DSL broadband connection in secure IP Sec-compliant VPNtechnology. A command line interface (CLI) can be used.

This application is related to copending patent applications entitled,“SYSTEM AND METHOD FOR CALL TRANSFER WITHIN AN INTERNET PROTOCOLCOMMUNICATIONS NETWORK,” which is filed on the same date and by the sameassignee and inventors, the disclosure which is hereby incorporated byreference.

Many modifications and other embodiments of the invention will come tothe mind of one skilled in the art having the benefit of the teachingspresented in the foregoing descriptions and the associated drawings.Therefore, it is understood that the invention is not to be limited tothe specific embodiments disclosed, and that modifications andembodiments are intended to be included within the scope of the appendedclaims.

1. A method of communicating, comprising: establishing a SessionInitiation Protocol (SIP) communications session call between a SIPcaller and a communications server using a call setup SIP Invitecommand; queueing the call from the SIP caller at the communicationsserver to place the SIP caller on hold; sending a first SIP Invitecommand as a call to a dialable extension from a call agent station tothe communications server to dequeue the call of the SIP caller who ison hold; sending a second SIP invite command between the call agentstation and the communications server as an alert indicative to transferthe call from the SIP caller to the call agent station; and initiatingan SIP Refer command between the SIP caller and the communicationsserver and transferring the call to the call agent station to establisha communications session between the caller and the call agent stationwithout user interaction.
 2. The method according to claim 1, furthercomprising dequeuing the call by dialing an extension from an SIP phoneat the call agent station and sending the SIP Invite command as adequeuing operation corresponding to a Session Description Protocol(SDP) offer and negotiating new and current media parameters.
 3. Themethod according to claim 2, further comprising negotiating new andcurrent media parameters using an SIP Invite command offer without anySDP media parameters to gain the new and current media parameters. 4.The method according to claim 2, further comprising negotiating new andcurrent media parameters with SDP media parameters.
 5. The methodaccording to claim 1, further comprising transferring media to thecaller while the call is queued using the Real Time Transport Protocol(RTP).
 6. The method according to claim 1, further comprising formingthe call agent station as an SIP phone.
 7. The method according to claim1, further comprising forming the communications server as an AutomaticCall Distributor (ACD).
 8. A method of communicating, comprising:establishing a Session Initiation Protocol (SIP) communications sessioncall between a SIP caller and a communications server using a call setupInvite command; queueing the call from the SIP caller at thecommunications server to place the SIP caller on hold; dequeuing thecall by dialing an extension from an SIP phone at a call agent stationand sending an SIP Invite command as a dequeuing operation correspondingto a Session Description Protocol (SDP) offer; sending a second SIPinvite command between the call agent station and communications serveras an alert indicative to transfer the call from the SIP caller to thecall agent station and negotiating new and current media parameters; andtransferring the call to the call agent station based on the new andcurrent media parameters by initiating an SIP Refer command between theSIP caller and communications server to establish a communicationssession between the caller and call agent station without userinteraction.
 9. The method according to claim 8, further comprisingnegotiating new and current media parameters using an SIP Invite commandoffer without any SDP media parameters to gain the new and current mediaparameters.
 10. The method according to claim 8, further comprisingnegotiating new and current media parameters with SDP media parameters.11. The method according to claim 8, further comprising transferringmedia to the caller while the call is queued using the Real TimeTransport Protocol (RTP).
 12. The method according to claim 8, furthercomprising forming the communications server as an Automatic CallDistributor (ACD).
 13. A communications system, comprising: a call agentstation configured to communicate in accordance with the SessionInitiation Protocol (SIP); and a communications server connected to saidcall agent station, and further comprising a queuing module, processorand transceiver configured to transmit and receive packet communicationsin accordance with the Session Initiation Protocol (SIP), said serverconfigured to receive from a SIP caller a call setup SIP Invite commandand establish a SIP call and queue the call within the queuing modulewhile placing the SIP caller on hold and waiting for live support from acall agent station to take the call, wherein said communications serveris configured to receive a first SIP Invite command from the call agentstation as a dialed extension to dequeue the call of the SIP caller whois on hold, and communicate a second SIP Invite command with the callagent station as an alert indicative to transfer the call from the SIPcaller to the call agent station and to initiate an SIP Refer commandwith the SIP caller to transfer the SIP call and establish acommunications session between the SIP caller and the call agent stationwithout user intervention.
 14. The communications system according toclaim 13, wherein said communications server and call agent station areconfigured to dequeue the call with a dialable extension from an SIPphone at the call agent station and as a dequeue operation correspondingto a Session Description Protocol (SDP) offer as part of an SIP Inviteto negotiate the new and current media parameters.
 15. Thecommunications system according to claim 14, wherein communicationsserver and call agent station are configured to negotiate new andcurrent media parameters using an SIP Invite command offer without anySDP media parameters to gain the new and current media parameters. 16.The communications system according to claim 14, wherein saidcommunications server and call agent station are configured to negotiatenew and current media parameters with SDP media parameters.
 17. Thecommunications system according to claim 13, wherein said communicationsserver is configured to transfer media to the caller while the call isqueued using the Real Time Transport Protocol (RTP).
 18. Thecommunications system according to claim 13, wherein said call agentstation comprises an SIP device phone.
 19. The communications systemaccording to claim 13, wherein said communications server comprises anAutomatic Call Distributor (ACD).